JP3345557B2 - Digital audio broadcast receiving apparatus and receiving method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To output an good audio signal, even when reception state has deteriorated. SOLUTION: An allocation/scale factor replacement section 15f counts number TH of times of error detection by a selection coding section 15b and stores allocation data and a scale factor included in an MPEG audio bit stream when the reception state is good while the TH is less than a 1st setting value TH1, and uses the stored allocation data and scale factor to replace those in the MPEG audio bit stream from the stored data, when the error detection number TH is more than the 1st setting value TH1 and the reception state is deteriorated, and an MPEG audio decoding section 16 decodes the replaced MPEG audio bit stream into a PCM audio signal.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital audio broadcast receiving apparatus and method, and more particularly, to a digital audio broadcast signal receiving apparatus , demodulating an MPEG audio bit stream from a received signal, and converting the MPEG audio bit stream into a digital audio broadcast signal. The present invention relates to a digital audio broadcast receiving apparatus and a receiving method for decoding audio sampling data into an audio signal by using included allocation data and a scale factor.

[0002]

2. Description of the Related Art Digital audio is now widely recognized for its excellent characteristics and ease of use, and is rapidly becoming popular. Against this background, digitalization of audio broadcasting has become active and digital audio broadcasting DAB (Digital Audio Broadcasting) has been realized in Europe. Digital audio broadcasting DAB
Performs orthogonal modulation on a frequency multiplexed broadcast signal including a plurality of data fields in which audio signal data of a predetermined program is inserted by frequency multiplexing and an information field in which program sequence data indicating the correspondence between the program and the data field is inserted. Transmit from a broadcasting station, receive the frequency multiplex broadcast signal in a frequency multiplex broadcast receiver, demodulate the frequency multiplex signal inserted in the data field according to the program separately specified by the user, convert it to an audio signal Output.

FIG. 4 shows a mode II frame structure employed in a DAB system. One frame length is 24 ms.
And a 2-symbol synchronization section SYNC and a 3-symbol high-speed information channel section FIC (Fast Information Channel).
l) and a data field part DFL of 72 symbols. The synchronization unit SYNC includes a null symbol and a synchronization information symbol used for recognizing the start point of the frame. In the high-speed information channel section FIC, time /
Date data, program arrangement data, service (program) name (label), program identification code SId (Service Identi
fication Code) is inserted, and a CRC for error correction is added at the end. Data field part DF
L is divided into 72 data fields, and a broadcast signal of a predetermined program is inserted into each data field by frequency division multiplexing, so that 6 to 8 programs can be transmitted simultaneously from one broadcasting station. . Which program corresponds to which data field is specified by the array data of the program included in the high-speed information channel unit FIC. FIG.
In the above, the frame structure of mode II for satellite broadcasting using the 1.5 GHz band was shown, but the frame structure of mode I using the frequency band of 50 MHz to 250 MHz for terrestrial broadcasting has almost the same structure, and one frame length is It is 96 ms.

FIG. 5 is a diagram showing the configuration of a DAB transmitter.
~ 1m is analog audio signal (audio signal of each program)
Is a data compression unit that performs high-efficiency encoding (data compression) on MPEG by MPEG. 2a to 2m are transmission path coding units,
A unit 3 for adding an error / detection correction code is a multiplexing unit, and each of the transmission line coding units 2a to 2 according to the program sequence data.
m output data in a predetermined data field (symbol)
And time division multiplexing. Reference numeral 4 denotes an interleave / OFDM modulator, and the modulation is performed by orthogonal frequency division multiplexing OFDM (Orthogonal Frequency Division MDM).
utiplex), and interleaving is performed by rearranging the order of multiplexed signals in a symbol according to a predetermined format. 5 is interleave / OFDM
I (In-phase) and Q (Quadrature) output from the modulator
Quadrature modulator for orthogonal transformation, respectively reference carrier and reference carrier to signal by multiplying the 90 ⁰ phase-shifted signals, 50～250MHz band transmission carrier wave obtained by the quadrature modulation 6 (mode I
In the case of (1), a frequency converter for converting into a high frequency, and 7 is a transmission power amplifier.

[0005] The MPEG audio compression system has high quality,
This is an ISO / IEC standard for high-efficiency stereo audio coding, uses 32-bit sub-band coding (band division coding) for compression, and achieves high-efficiency compression using psychoacoustic characteristics. I do. The human ear cannot hear sounds below a certain level, and a characteristic curve obtained by plotting this level for each band is called a minimum masking threshold curve (minimum audible limit curve) MTC (FIG. 6). The masking effect changes depending on the situation of surrounding sounds, and even a sound having a level equal to or higher than the minimum masking threshold curve MTC cannot hear a small sound due to a loud sound. This is because the loud sound changes the masking threshold curve like MTC 'in FIG. 6, and the sound components below the curve are masked and cannot be heard by human ears. In consideration of the above, sounds below the masking threshold level MTC 'are not quantized, but sounds above the masking threshold level are quantized. When quantizing, quantization is performed by assigning the number of quantization bits according to the difference between the audio level and the masking threshold level in each subband, and the quantized data (sampling audio data) and the assigned number of bits are assigned. Etc. are output.

More specifically, an audio signal is subdivided into 32 bands, and subband encoding of 32 bands is performed. That is, the entire band is divided into 32 equally spaced frequency widths, and each signal is sampled and encoded at 1/32 of the original sampling period. In MPEG audio,
Encoding is performed by converting 384 (= 12 × 32) samples into one frame, converting each subband into 12 samples and 32 bands. Data of 12 samples in one band is divided into an L channel (0 channel) waveform and an R channel (1 channel) waveform, and a scale factor is determined for each. That is, normalization is performed so that the maximum value of each of the twelve waveforms becomes 1.0, and the normalized magnification is encoded as a scale factor. By this encoding, the generation of quantization noise can be limited, the effect of psychoacoustics works, and the noise is hardly detected.

[0007] The number of quantization bits for each channel of each subband is determined and used as allocation data. The masking effect can be used most effectively by specifying the quantization precision (the number of quantization bits) up to the last masking level. As a result of the masking, a band containing only a signal of a level not recognized by the auditory system can completely lose information. In such a case, no bit is assigned as sample data (no sampling data). The allocation data indicates the number of quantization bits of the sample data of each channel (0 channel, 1 channel) in each subband. When the number of quantization bits = 0, there is no sampling data.

FIG. 7 is a diagram for explaining the structure of one frame of an MPEG audio bit stream.
o Access Unit). AAU is one by one
Is the minimum unit that can be decoded into an audio signal, and always contains data of a fixed number of samples = 384 samples.
The AAU includes a 32-bit header section 1, an error check code (optional) 2, and an audio data section 3, and the audio data section 3 includes allocation data 3
a, a scale factor 3b, and sample data 3c. The header section 1 includes a 12-bit all "1" synchronization word 1a, an ID "1b" that is always "1", other layers 1c, a bit rate index, a sampling frequency, mode, and other information.

The audio data section 3 has a structure as shown in FIG. That is, allocation data 3
a is each channel ch of each subband sb (0 to 31)
(0 channel, 1 channel) indicates the number of quantization bits of 12 pieces of sampling data, and scale factor 3b indicates the normalization magnification of each of the allocation bits other than 0. Each sampled data of the sub-band Sb and the channel ch whose allocation bit number is not 0 is multiplied by the corresponding scale factor Sij and quantized by the allocation bit number to become sample data 3c.

FIG. 9 is a block diagram of a DAB receiver (frequency multiplex broadcast receiver). 11 denotes an antenna, 12 is a front-end, 13 quadrature demodulator for quadrature detection by multiplying each of the intermediate frequency signal to the reference carrier and reference carrier 90 ⁰ phase-shifted signal outputted from the front end, 14 is a quadrature demodulator A / D converter for AD-converting the baseband frequency multiplexed signal output from the A / D converter at a predetermined sampling frequency, and a transmission path 15 for performing FFT demodulation processing, deinterleaving, decoding processing and the like to restore / output the MPEG audio bit stream STM Decoding circuit, 16 is MP
EG audio bitstream STM to original PCM
An MPEG audio decoding unit 17 for decoding audio data is a DA converter for converting PCM audio data into an analog audio signal.

The transmission line decoding circuit 15 includes an FFT differential demodulation unit 15a, a selective decoding unit 15b, a deinterleave unit 15
c, a control microcomputer 15d, and a synchronization processor 15e. The FFT differential demodulation unit 15a performs an FFT process on the digital data output from the AD converter 14, identifies a frequency included in the frequency multiplexed signal of each symbol, and demodulates interleaved transmission data based on the identified frequency. . Deinterleave 15c is FF
The interleave is decomposed from the output data of the T differential demodulation unit to return to the original data sequence, and the selective decoding unit 15b performs error detection and correction on the deinterleaved data and inputs the data to the control microcomputer 15d. The control microcomputer 15d detects the symbol position (data field) of the program specified by the user based on the program sequence data included in the high-speed information channel unit FIC, and notifies the selective decoding unit 15b.

The selective decoding unit 15b outputs decoded data (MPEG audio bit stream STM) of the notified symbol (data field) to the MPEG audio decoding unit 16. MPEG audio decoding unit 1
Numeral 6 converts the input MPEG audio bit stream STM into the original PCM audio data and outputs it. That is, the MPEG audio decoding unit 16 decodes the data into PCM audio data using the number of allocation bits, the scale factor, and the sample data.
The M audio data is input to the DA converter 17. or,
The MPEG audio decoding unit 16 outputs a data string PAD (Program) related to the program contents included in the audio data.
associated data), for example, information such as the artist's name, music title, and lyrics are sent to the control microcomputer 15d. The control microcomputer 15d displays the input information on the display unit of the operation unit via the user interface. or,
The control microcomputer 15d sends the label (program name) included in the high-speed information channel FIC to the operation unit as appropriate according to the user's request, and displays it. Further, the control microcomputer 15d
Also performs the tuning control of the DAB station.

The synchronization processor 15e performs FFT processing on the synchronization symbol after detecting the null symbol, calculates the frequency offset of the input signal, and converts the AFC signal into the orthogonal demodulator 13
To adjust the reference frequency. Synchronous processor 1
5e calculates the correlation (shift amount) between the FFT result of the synchronization symbol and the original synchronization symbol, and calculates the result as the inverse FF.
T processing and the impulse response CIR (channel Imp
ulse Response) is output. Front end 12 is CI
The reference clock generator of the receiver, that is, the oscillation frequency of the voltage controlled crystal oscillator is synchronized with the transmission frequency of the transmitter based on the R signal.

[0014]

A CN ratio (Carrier to Noise ratio) which is a ratio between a carrier signal and a noise signal contained therein.
Is deteriorated, that is, when the reception condition is deteriorated, MP
The synchronization word in the EG audio bit stream cannot be detected, or the number of allocation bits or the scale factor is incorrectly detected. If the sync word cannot be detected, the audio signal cannot be correctly decoded, and noise occurs. Further, when a scale factor detection error occurs, a small sound is decoded into a loud sound to generate noise, and when a detection error of the allocation bit number occurs, quantization noise occurs. In view of the above, it is an object of the present invention to be able to output a good sound even when the reception state (C / N ratio) is deteriorated . Another object of the present invention is to mute the audio when the reception condition is further deteriorated, or to receive and output another DAB station.
Is to do so.

[0015]

According to the present invention, there is provided an MPEG audio bit stream receiving a digital audio broadcast signal to which an error detection and correction code is added, performing error detection and correction processing on the received signal. A digital audio broadcast receiving method for demodulating the audio data and decoding the audio sampling data into an audio signal using the allocation data and the scale factor included in the MPEG audio bit stream. Means for storing allocation data and a scale factor included in an MPEG audio bitstream when the reception condition is equal to or less than 1 and the number of error detections is equal to or greater than the first set value and the reception condition is degraded Time, memorized allocation MPE using the scale factor
This is achieved by means of decoding the G audio bitstream into an audio signal. Further, according to the present invention, when the reception state further deteriorates and the number of times of error detection becomes equal to or more than the second set value, the audio output is switched to a unit for muting the audio output or to reception of another digital broadcasting station. Achieved by means.

[0016]

BEST MODE FOR CARRYING OUT THE INVENTION

(A) Overall Configuration FIG. 1 is a configuration diagram of an in-vehicle digital broadcast receiver according to the present invention, and the same parts as those of the conventional digital broadcast receiver of FIG. In the figure, 11 denotes an antenna, 12 is a front-end, 13 quadrature demodulator for quadrature detection by multiplying the intermediate frequency signal output from the front end reference carrier and the reference carrier 90 ⁰ phase-shifted signals, respectively,
Reference numeral 14 denotes an A that converts the baseband frequency multiplexed signal output from the quadrature demodulator into an analog signal at a predetermined sampling frequency.
A D converter 15 is a transmission line decoding circuit that performs FFT demodulation processing, deinterleaving, decoding processing and the like to restore / output an MPEG audio bit stream STM.
6 is an MPEG audio decoder for decoding the MPEG audio bit stream STM into the original PCM audio data, 17 is a DA converter which converts the PCM audio data to an analog audio signal, 18 is zero audio output based on the mute command MUT. And a mute circuit for applying an audio mute.

The transmission line decoding circuit 15 includes an FFT differential demodulation unit 15a, a selective decoding unit 15b, a deinterleave unit 15
c, control microcomputer 15d, synchronization processor 15e,
An allocation / scale factor changing unit 15f is provided. The FFT differential demodulation unit 15a includes an AD converter 14
Performs FFT processing on the digital data output from, identifies the frequency included in the frequency multiplexed signal of each symbol,
Demodulate the interleaved transmission data based on the identified frequency. The deinterleaving unit 15c deinterleaves the output data of the FFT differential demodulation unit to return to the original data sequence, and the selective decoding unit 15b performs error detection and correction processing on the deinterleaved data and controls the high-speed information channel unit FIC. Data of the symbol position (data field) of the program designated by the user (MPEG audio bit stream S
TM) is input to the allocation / scale factor changing section 15f at the next stage. The selective decoding section 15b outputs one error detection signal EDS every time an error is detected by the error detection and correction processing.
Input to 5f.

The control microcomputer 15d detects the symbol position (data field) of the program designated by the user based on the program sequence data included in the high-speed information channel unit FIC, notifies the selected symbol position (data field) to the selective decoding unit 15b. Tuning control for selecting a designated digital broadcasting station (DAB station) is performed. Note that the selective decoding unit 15b uses the symbol (data field) notified from the control microcomputer 15d.
The decoded data (MPEG audio bit stream STM) of the allocation / scale factor changing unit 1
Input to 5f.

The allocation / scale factor changing section 15f counts the error detection signal EDS, and detects the reception state (C) where the number of error detections per predetermined time is equal to or less than a first set value.
/ N ratio) is good, the MPEG audio bit stream STM is directly used by the MPEG audio decoding unit 1
6 and stores / saves allocation data and scale factors contained in the MPEG audio bitstream STM. Allocation /
When the number of error detections per predetermined time is equal to or greater than the first set value and the reception condition is deteriorated, the scale factor reassigning unit 15f converts the stored MPEG audio bit stream received with the allocation and scale factor into a bad one. The allocation data and the scale factor included are changed, and the result is output to the MPEG audio decoding unit 16. Further, the allocation / scale factor changing unit 15f outputs a mute command MUT for muting the audio output when the reception state is further deteriorated and the number of error detections per predetermined time is equal to or more than a second set value. Alternatively, D for switching to reception of another digital broadcast station
An AB station switching command STS is output.

That is, as shown in FIG. 2, when the number of error detections TH per predetermined time is in a good reception state RC1 equal to or less than the first set value TH1, the allocation / scale factor changing section 15f outputs the MPEG audio bit. Store / save the allocation data and scale factor contained in the stream STM. In addition, the number of error detections TH per predetermined time is reduced to a first set value TH due to deterioration of the reception state.
When the value becomes 1 or more (reception state RC2), the allocation data and the scale factor included in the MPEG audio bit stream received with the stored allocation and scale factor are replaced and output. Further, the number of error detections per predetermined time T
H is greater than or equal to the second set value TH2 (in the receiving state RC
3), mute command MU to mute audio output
T or a DAB station switching command STS for switching to reception of another digital broadcasting station. It should be noted here that even if the reception state deteriorates considerably and the number of error detections TH becomes equal to or greater than TH2 ', muting is not performed and allocation and scale factor replacement are continued. This is because DAB broadcasting assumes a terrestrial broadcasting station, so that even if the receiving condition deteriorates due to the running of the vehicle, there is a high possibility that the receiving condition will be recovered immediately. This is because it is more effective to output the noise if the noise does not cause a problem in hearing, rather than completely cut off the sound. In addition, TH>
Even at the time TH2 ', since the allocation and the scale factor are changed, the noise is reduced and no problem occurs in the sense of hearing.

The MPEG audio decoder 16 converts the input MPEG audio bit stream STM into the original PCM audio data and outputs it. That is, the MPEG audio decoding unit 16 decodes the PCM audio data using the number of allocation bits, the scale factor, and the sample data, and inputs the PCM audio data to the DA converter 17. Further, the MPEG audio decoding unit 16 sends information such as the artist name, music title, and lyrics related to the program contents included in the audio data to the control microcomputer 15d. The control microcomputer 15d displays the input information on the display unit of the operation unit via the user interface.

The synchronization processor 15e performs FFT processing on the synchronization symbol after detecting the null symbol, calculates the frequency offset of the input signal, and converts the AFC signal into the orthogonal demodulator 13
To adjust the reference frequency. Synchronous processor 1
5e calculates the correlation (shift amount) between the FFT result of the synchronization symbol and the original synchronization symbol, and calculates the result as the inverse FF.
T processing and the impulse response CIR (channel Imp
ulse Response) is output. Front end 12 is CI
The reference clock generator of the receiver, that is, the oscillation frequency of the voltage controlled crystal oscillator is synchronized with the transmission frequency of the transmitter based on the R signal.

(B) Allocation / scale factor changing section (b-1) Configuration FIG. 3 shows an allocation / scale factor changing section 15f.
61 is a configuration diagram of a synchronization detection unit 61 for detecting a synchronization word from the MPEG audio bit stream STM;
Reference numeral 62 denotes an intra-frame bit position output unit having a built-in counter. The counter has the same capacity as the number of AAU bits in one frame, is set to a predetermined value (for example, 0) by detecting a synchronization word, counts a bit clock BCL, and monitors a bit position in a frame based on the count value. ,Output. Reference numeral 63 denotes a header detection / storage unit for detecting and storing a header (excluding a sync word / CD) from the bit stream STM, 64 an allocation detection / storage unit for detecting and storing allocation data, and 65 a scale factor detection. A scale factor detection / storage section 66 stores and stores a 12-bit sync word of all "1" and 1
This is a synchronization word / ID generation unit that generates an ID of bit “1”.

Reference numeral 67 denotes a control unit having a microcomputer configuration, reference numeral 68 denotes a switching circuit, reference numeral 69 denotes a counter for counting the error detection signal EDS output from the selective encoding unit 15b, reference numeral 70 denotes the number of error detections TH per predetermined time and a set value TH1. , TH2 (see FIG. 2). Control unit 67
When the reception state (C / N ratio) deteriorates and the number of error detections TH per predetermined time becomes equal to or more than the first set value TH1, the switching circuit 68 is controlled to control the synchronization word / ID, header, allocation data, scale, Controls factor replacement. That is, when the reception state is good, the control unit 67 sets M
The switching circuit 68 is instructed to select the PEG audio bit stream STM. However, the reception condition deteriorates and T
When H> TH1, the control unit 67 changes the synchronization word / ID, header, allocation data, and scale factor stored in each of the units 63 to 66 based on the bit position output from the intra-frame bit position output unit 62. The timing signals are sequentially input to the switching circuit 68. or,
When the reception state further deteriorates and TH> TH2, a mute command MUT is output, and the mute circuit 18 mutes the sound. When TH> TH1, a DAB station switching command STS is output instead of muting, and the user tunes / receives to the DAB station preset in the control microcomputer 15d (FIG. 1) from the operation unit by the user. It can also be configured as follows.

When the reception state is good and the selection of the bit stream STM is instructed, the switching circuit 68 inputs the bit stream STM to the MPEG audio decoding unit 69 as it is. However, the reception condition deteriorates and TH
> TH1, and when the data replacement timing signal is input from the control section 67, the switching circuit 68 selects and outputs the latest corresponding data detected and stored when the reception state is good. That is, the switching circuit 68
When the synchronization word / ID replacement timing signal is input from the control unit 67, the synchronization word / ID in the bit stream is replaced with the synchronization word / ID output from the synchronization word / ID generation unit 66. Also, the switching circuit 68
When the header replacement timing signal is input from the control unit 67, the header in the bit stream is replaced with the header (excluding the synchronization word / ID) stored in the header detection / storage unit 63, and the allocation replacement timing signal is transmitted to the control unit 67. When input from the allocation unit 67, the allocation data in the bit stream is replaced with the allocation data stored in the allocation detection / storage unit 64. When a scale factor replacement timing signal is input from the control unit 67, the scale factor detection / storage unit The scale factor of the bit stream is replaced with the scale factor stored in 65. Note that the sample data in the bit stream STM is
Input to the G audio decoding unit 16.

The MPEG audio decoding unit 16 synchronizes upon detection of a synchronization word, and thereafter extracts allocation data (the number of allocation bits), a scale factor, and sample data from the bit stream.
From these, the PCM audio data is restored, and the DA converter 17 DA-converts the PCM audio data and outputs it.

(B-2) Operation When the reception state is good, the MPEG audio decoding unit 1
6 is an MPE output from the selection encoding unit 15b (FIG. 1).
The sync word / ID, header, allocation data, scale factor, and sample data can be correctly detected from the G audio bit stream STM, and the PCM audio data is decoded and output using these. However, if the receiving condition deteriorates, the sync word / ID, header,
Allocation data and scale factor cannot be detected correctly, and large noise is generated.

Therefore, the allocation / scale factor changing section 15f operates each detection / storage section 6 when the reception state is good.
These data are detected and stored in 3-65. Then, when the reception state deteriorates and TH> TH1, the synchronization word / ID, the header, the allocation data, and the scale factor in the bit stream STM are detected in the fixed synchronization word / ID and when reception is good, respectively. Replace with the stored header, allocation data, and scale factor. When the reception state deteriorates, the MPEG audio decoding unit 16 detects the replaced sync word / ID, allocation data (allocation bit number), scale factor, and sample data from the beat stream, and uses these to detect the PCM audio data. And the DA converter 17 converts the PCM audio data into an analog audio signal and outputs the analog audio signal.

In this way, even if the reception state deteriorates, the MPEG audio decoding unit 16 can reliably detect the synchronization word / ID, and furthermore, the allocation data (the number of allocation bits) and the reception state as a scale factor. The PCM audio data can be restored using the immediately preceding value with good, and noise can be reduced. If the receiving condition is further deteriorated and TH> TH2, the control unit 67 outputs a mute signal MUT to mute the sound, or sets the mute signal MUT in advance or sets it in the control microcomputer 15d different from the DAB station currently receiving. And receives and outputs the broadcast of the DAB station. In addition, DA
Since the B broadcast is assumed to be a terrestrial broadcast station, the probability of being able to receive a broadcast from another receive station even when the current receive station cannot be received is much higher than when a satellite is used. As described above, the present invention has been described with reference to the embodiments. However, the present invention can be variously modified in accordance with the gist of the present invention described in the claims, and the present invention does not exclude these.

[0030]

As described above, according to the present invention, the number of error detections is counted, and the allocation data and the scale factor included in the MPEG audio bit stream when the number of error detections is equal to or less than the first set value and the reception state is good. Is stored, and when the number of times of error detection becomes equal to or more than the first set value and the reception condition deteriorates, the MPEG audio bit stream is decoded into an audio signal using the stored allocation and scale factor. Therefore, even if the C / N ratio is low, it is possible to output a good sound while suppressing noise. Further, according to the present invention, when the C / N ratio further deteriorates and the number of times of error detection becomes equal to or more than the second set value, the audio output is muted or the reception is switched to another digital broadcasting station. Therefore, the generation of noise can be prevented, or the broadcast of another good DAB station can be received and output.

[Brief description of the drawings]

FIG. 1 is a configuration diagram of a digital broadcast receiver of the present invention.

FIG. 2 is an explanatory diagram of an operation of an allocation / scale factor changing unit.

FIG. 3 is a configuration diagram of an allocation / scale factor changing unit.

FIG. 4 is a diagram illustrating the structure of a DAB signal frame.

FIG. 5 is a configuration diagram of a DAB transmitter.

FIG. 6 is a masking threshold characteristic diagram.

FIG. 7 is a diagram illustrating the structure of an AAU.

FIG. 8 is a configuration diagram of an audio data section in the AAU.

FIG. 9 is a configuration diagram of a conventional DAB receiver.

[Explanation of symbols]

 15. Transmission line decoding unit 15b Selection encoding unit 15d Control microcomputer 15f Allocation / scale factor changing unit 16 MPEG audio decoding unit 18 Mute circuit

──────────────────────────────────────────────────続 き Continuation of the front page (56) References JP-A-7-336311 (JP, A) JP-A-8-65275 (JP, A) JP-A-62-117423 (JP, A) 116314 (JP, A) JP-A-6-104850 (JP, A) JP-A-8-242215 (JP, A) JP-A-7-86963 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) H04L 1/00 G10L 13/00 H04B 14/04 H04H 1/00

Claims (4)

(57) [Claims] 1. A digital audio broadcast signal to which an error detection / correction code has been added is received, an error detection / correction process is performed on the received signal, and an MPEG audio bit stream is demodulated and included in the MPEG audio bit stream. A digital audio broadcast receiving method for decoding audio sampling data into an audio signal using allocation data and a scale factor, wherein the number of error detections is counted, and the number of error detections is equal to or less than a first set value. Allocation data and a scale factor included in an audio bit stream are stored, and when the number of error detections exceeds a first set value and reception conditions deteriorate, MPEG audio is stored using the stored allocation data and scale factor.・ Bit A digital audio broadcast receiving method, wherein the digital stream is decoded into an audio signal. 2. The digital audio broadcast according to claim 1, wherein when the number of times of error detection becomes equal to or greater than a second set value, the audio output is muted or the reception is switched to another digital broadcast station. Receiving method. 3. A digital audio broadcast signal to which an error detection and correction code is added is received, an error detection and correction process is performed on the received signal to demodulate an MPEG audio bit stream, and the demodulated MPEG audio bit stream is included in the MPEG audio bit stream. In a digital audio broadcast receiving apparatus that decodes audio sampling data into an audio signal using allocation data and a scale factor, an error detection / correction unit that performs error detection / correction processing on the received signal, and the number of error detections detected by the error detection / correction processing. A counting unit that counts, a comparing unit that compares the number of error detections with a set value, and allocation data and a scale factor included in the MPEG audio bitstream when the number of error detections is equal to or less than a first set value and the reception state is good. Memory part to memorize, mistake An MPEG audio decoding unit that decodes an MPEG audio bit stream into an audio signal using the stored allocation data and the scale factor when the number of times of output becomes equal to or greater than a first set value and the reception condition deteriorates. A digital audio broadcast receiver. Wherein when the error detection count is equal to or higher than the second set value, claim mutes the audio output, or characterized by comprising a control unit, to switch the reception of another digital broadcasting station 3. The digital audio broadcast receiving device according to 3.